This application is a continuation of prior U.S. Ser. No. 09/606,824, filed Jun. 28, 2000, now abandoned entitled “METHOD AND APPARATUS FOR CALL SETUP WITHIN A VOICE FRAME NETWORK”.
This invention relates to Internet protocol (IP) network systems in which voice or other data are sent in packets from a server to a client or vice versa. More specifically, it concerns method and apparatus for endpoint source and destination call setup within a voice over Internet protocol (VoIP) or similar voice frame network.
Detecting, and optionally routing around, voice frame network congestion typically is performed by a method that is similar to Internet protocol (IP) pinging. Those of skill in the art will appreciate that pinging is a method by which real-time responder (RTR) or service assurance agent (SAA) endpoint probes are transmitted and echoed across the network at call setup time to determine the inter-connective preparedness of endpoints for incoming calls.
A network endpoints coordination protocol (NECP) may measure network latency between network endpoints, as described in co-pending U.S. patent application Ser. No. 09/346,080 entitled A PROTOCOL TO COORDINATE NETWORK END POINTS TO MEASURE NETWORK LATENCY, filed Jul. 1, 1999, the disclosure of which is incorporated herein by reference. Network endpoints metrics such as data packet delay, jitter and loss may be obtained, as described in co-pending U.S. patent application Ser. No. 09/434,845 entitled METHOD AND APPARATUS FOR MEASURING NETWORK DATA PACKET DELAY, JITTER AND LOSS, filed Nov. 4, 1999, the disclosure of which also is incorporated herein by reference. These protocol, method and apparatus rely on RTR/SAA endpoint probe transmission and echoing with or without content modification. Both patent applications are subject to common ownership herewith by assignee Cisco Technology, Inc.
One recently proposed VoIP congestion detection method permits the endpoint probe to emulate a burst of actual voice packets. This provides improved fidelity to the probe results compared with a simple IP ping. The mechanism involves sending an RTR/SAA probe to the IP address of the far-end voice gateway prior to attempting a call setup to the gateway. To increase efficiency, and to reduce post-dialing delay, the voice gateway that originates the call keeps a cache of recent RTR/SAA probe results.
The cache allows the gateway to consolidate probe requests from plural voice ports and associated call attempts, so that a single probe test and results is shared by all voice ports and call attempts on a voice gateway. This reduces the volume of IP network traffic and congestion generated by the RTR/SAA probes themselves. This is particularly useful in conjunction with network maintenance techniques that continuously send probes to a far-end gateway to confirm connectivity on a pre-emptive basis, rather than simply at call setup time.
This probe result caching scheme suffers from the limitation that the number of IP addresses available as targets for VoIP calls is likely to be larger than any reasonable RTR/SAA cache size. The cache entries typically store an RTR/SAA probe result for each IP address corresponding with each distant voice gateway or VoIP endpoint. When this mechanism is used with an IP private branch exchange (PBX), each individual IP phone as part of the IP PBX may have an individual IP address. Accordingly, each individual IP phone may contribute an individual IP probe cache entry, which requires an unreasonable amount of caching time and resource, e.g. memory.
Moreover, not all IP targets or VoIP endpoints support RTR/SAA probe/response queries. In these cases, it is not possible to perform either initial or periodic connectivity confirmation.